Merge AudioRendererImpl and AudioRendererBase; add NullAudioSink

content/renderer/media/audio_renderer_impl.cc

-// Copyright (c) 2012 The Chromium Authors. All rights reserved.
-// Use of this source code is governed by a BSD-style license that can be
-// found in the LICENSE file.
-
-#include "content/renderer/media/audio_renderer_impl.h"
-
-#include <math.h>
-
-#include <algorithm>
-
-#include "base/bind.h"
-#include "content/common/child_process.h"
-#include "content/common/media/audio_messages.h"
-#include "content/renderer/media/audio_hardware.h"
-#include "content/renderer/render_thread_impl.h"
-#include "media/audio/audio_buffers_state.h"
-#include "media/audio/audio_util.h"
-#include "media/base/filter_host.h"
-
-AudioRendererImpl::AudioRendererImpl(media::AudioRendererSink* sink)
-    : AudioRendererBase(),
-      bytes_per_second_(0),
-      stopped_(false),
-      sink_(sink),
-      is_initialized_(false) {
-}
-
-AudioRendererImpl::~AudioRendererImpl() {
-}
-
-base::TimeDelta AudioRendererImpl::ConvertToDuration(int bytes) {
-  if (bytes_per_second_) {
-    return base::TimeDelta::FromMicroseconds(
-        base::Time::kMicrosecondsPerSecond * bytes / bytes_per_second_);
-  }
-  return base::TimeDelta();
-}
-
-void AudioRendererImpl::UpdateEarliestEndTime(int bytes_filled,
-                                              base::TimeDelta request_delay,
-                                              base::Time time_now) {
-  if (bytes_filled != 0) {
-    base::TimeDelta predicted_play_time = ConvertToDuration(bytes_filled);
-    float playback_rate = GetPlaybackRate();
-    if (playback_rate != 1.0f) {
-      predicted_play_time = base::TimeDelta::FromMicroseconds(
-          static_cast<int64>(ceil(predicted_play_time.InMicroseconds() *
-                                  playback_rate)));
-    }
-    earliest_end_time_ =
-        std::max(earliest_end_time_,
-                 time_now + request_delay + predicted_play_time);
-  }
-}
-
-bool AudioRendererImpl::OnInitialize(int bits_per_channel,
-                                     ChannelLayout channel_layout,
-                                     int sample_rate) {
-  // We use the AUDIO_PCM_LINEAR flag because AUDIO_PCM_LOW_LATENCY
-  // does not currently support all the sample-rates that we require.
-  // Please see: http://code.google.com/p/chromium/issues/detail?id=103627
-  // for more details.
-  audio_parameters_.Reset(
-      media::AudioParameters::AUDIO_PCM_LINEAR,
-      channel_layout, sample_rate, bits_per_channel,
-      audio_hardware::GetHighLatencyOutputBufferSize(sample_rate));
-
-  bytes_per_second_ = audio_parameters_.GetBytesPerSecond();
-
-  DCHECK(sink_.get());
-
-  if (!is_initialized_) {
-    sink_->Initialize(audio_parameters_, this);
-
-    sink_->Start();
-    is_initialized_ = true;
-    return true;
-  }
-
-  return false;
-}
-
-void AudioRendererImpl::OnStop() {
-  if (stopped_)
-    return;
-
-  DCHECK(sink_.get());
-  sink_->Stop();
-
-  stopped_ = true;
-}
-
-void AudioRendererImpl::SetPlaybackRate(float rate) {
-  DCHECK_LE(0.0f, rate);
-
-  // Handle the case where we stopped due to IO message loop dying.
-  if (stopped_) {
-    AudioRendererBase::SetPlaybackRate(rate);
-    return;
-  }
-
-  // We have two cases here:
-  // Play: GetPlaybackRate() == 0.0 && rate != 0.0
-  // Pause: GetPlaybackRate() != 0.0 && rate == 0.0
-  if (GetPlaybackRate() == 0.0f && rate != 0.0f) {
-    DoPlay();
-  } else if (GetPlaybackRate() != 0.0f && rate == 0.0f) {
-    // Pause is easy, we can always pause.
-    DoPause();
-  }
-  AudioRendererBase::SetPlaybackRate(rate);
-}
-
-void AudioRendererImpl::Pause(const base::Closure& callback) {
-  AudioRendererBase::Pause(callback);
-  if (stopped_)
-    return;
-
-  DoPause();
-}
-
-void AudioRendererImpl::Seek(base::TimeDelta time,
-                             const media::PipelineStatusCB& cb) {
-  AudioRendererBase::Seek(time, cb);
-  if (stopped_)
-    return;
-
-  DoSeek();
-}
-
-void AudioRendererImpl::Play(const base::Closure& callback) {
-  AudioRendererBase::Play(callback);
-  if (stopped_)
-    return;
-
-  if (GetPlaybackRate() != 0.0f) {
-    DoPlay();
-  } else {
-    DoPause();
-  }
-}
-
-void AudioRendererImpl::SetVolume(float volume) {
-  if (stopped_)
-    return;
-  DCHECK(sink_.get());
-  sink_->SetVolume(volume);
-}
-
-void AudioRendererImpl::DoPlay() {
-  earliest_end_time_ = base::Time::Now();
-  DCHECK(sink_.get());
-  sink_->Play();
-}
-
-void AudioRendererImpl::DoPause() {
-  DCHECK(sink_.get());
-  sink_->Pause(false);
-}
-
-void AudioRendererImpl::DoSeek() {
-  earliest_end_time_ = base::Time::Now();
-
-  // Pause and flush the stream when we seek to a new location.
-  DCHECK(sink_.get());
-  sink_->Pause(true);
-}
-
-size_t AudioRendererImpl::Render(const std::vector<float*>& audio_data,
-                                 size_t number_of_frames,
-                                 size_t audio_delay_milliseconds) {
-  if (stopped_ || GetPlaybackRate() == 0.0f) {
-    // Output silence if stopped.
-    for (size_t i = 0; i < audio_data.size(); ++i)
-      memset(audio_data[i], 0, sizeof(float) * number_of_frames);
-    return 0;
-  }
-
-  // Adjust the playback delay.
-  base::TimeDelta request_delay =
-      base::TimeDelta::FromMilliseconds(audio_delay_milliseconds);
-
-  // Finally we need to adjust the delay according to playback rate.
-  if (GetPlaybackRate() != 1.0f) {
-    request_delay = base::TimeDelta::FromMicroseconds(
-        static_cast<int64>(ceil(request_delay.InMicroseconds() *
-                                GetPlaybackRate())));
-  }
-
-  int bytes_per_frame  = audio_parameters_.GetBytesPerFrame();
-
-  const size_t buf_size = number_of_frames * bytes_per_frame;
-  scoped_array<uint8> buf(new uint8[buf_size]);
-
-  uint32 frames_filled = FillBuffer(buf.get(), number_of_frames, request_delay);
-  uint32 bytes_filled = frames_filled * bytes_per_frame;
-  DCHECK_LE(bytes_filled, buf_size);
-  UpdateEarliestEndTime(bytes_filled, request_delay, base::Time::Now());
-
-  // Deinterleave each audio channel.
-  int channels = audio_data.size();
-  for (int channel_index = 0; channel_index < channels; ++channel_index) {
-    media::DeinterleaveAudioChannel(buf.get(),
-                                    audio_data[channel_index],
-                                    channels,
-                                    channel_index,
-                                    bytes_per_frame / channels,
-                                    frames_filled);
-
-    // If FillBuffer() didn't give us enough data then zero out the remainder.
-    if (frames_filled < number_of_frames) {
-      int frames_to_zero = number_of_frames - frames_filled;
-      memset(audio_data[channel_index] + frames_filled,
-             0,
-             sizeof(float) * frames_to_zero);
-    }
-  }
-  return frames_filled;
-}
-
-void AudioRendererImpl::OnRenderError() {
-  host()->DisableAudioRenderer();
-}
-
-void AudioRendererImpl::OnRenderEndOfStream() {
-  // TODO(enal): schedule callback instead of polling.
-  if (base::Time::Now() >= earliest_end_time_)
-    SignalEndOfStream();
-}